Adds Analog Gain Control (AGC) to the WebRTC client.

content/renderer/media/webrtc_audio_device_impl.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #pragma once
 
 #include <string>
 #include <vector>
@@ skipping 73 matching lines @@
 //                                .---------------------.
 //
 //  (*) Using SyncSocket for inter-process synchronization with low latency.
 //      The actual data is transferred via SharedMemory. IPC is not involved
 //      in the actual media transfer.
 //
 // Implementation notes:
 //
 //  - This class must be created on the main render thread.
 //  - The webrtc::AudioDeviceModule is reference counted.
-//  - Recording is currently not supported on Mac OS X.
+//  - Regarding the AGC: It aims at maintaining the same speech loudness level

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

s/It aims at maintaining/It aims to maintain

It's not very clear what "the same" refers to.  I'm assuming it means 'even out'
or normalize the volume, but there might be a clearer way to put it.

You could also just point to http://en.wikipedia.org/wiki/Automatic_gain_control
and skip to the implementation specific things.

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Modified to "It aims to maintain a constant speech loudness level from the
microphone." and added a reference to Wikipedia.

+//    from the microphone. This is done by both controlling the analog
+//    microphone gain and applying a digital gain. The microphone gain on the

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

s/a digital gain/digital gain

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

+//    sound card is slowly increased/decreased during speech only. By observing
+//    the microphone control slider you can see it move when you speak. If you
+//    scream, the slider moves downwards and then upwards again when you return
+//    to normal. It is not uncommon that the slider hits the maximum. This
+//    means that the maximum analog gain is not large enough to give the
+//    desired loudness. Nevertheless, we can in general still attain the
+//    desired loudness. If the microphone control slider is moved manually,
+//    the analog adaptation restarts and returns to roughly the same position

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

s/analog adaptation/gain adaptation.

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Actually, it is only the analog part that restarts; not the digital. Will change
as you propose.

+//    as before the change if the circumstances are still the same. When the
+//    input microphone signal causes saturation, the level is decreased
+//    dramatically and has to re-adapt towards the old level. The adaptation
+//    is a slowly varying process and at the beginning of a call this is

[tommi (sloooow) - chröme, 2012/03/26 15:26:40]

s/beginning of a call/beginning of capture
(there are no assumptions about a 'call' in the code afaics)

[henrika (OOO until Aug 14), 2012/03/27 09:20:38]

Done.

+//    noticed by a slow increase in volume. Smaller changes in microphone input
+//    level is leveled out by the built-in digital control. For larger
+//    differences we need to rely on the slow adaptation.
 //
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public webrtc::AudioDeviceModule),
       public media::AudioRendererSink::RenderCallback,
       public AudioInputDevice::CaptureCallback,
       public AudioInputDevice::CaptureEventHandler {
  public:
   // Methods called on main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
   virtual int32_t AddRef() OVERRIDE;
   virtual int32_t Release() OVERRIDE;
 
   // We need this one to support runnable method tasks.
   static bool ImplementsThreadSafeReferenceCounting() { return true; }
 
   // AudioDevice::RenderCallback implementation.
   virtual size_t Render(const std::vector<float*>& audio_data,
                         size_t number_of_frames,
                         size_t audio_delay_milliseconds) OVERRIDE;
   virtual void OnRenderError() OVERRIDE;
 
   // AudioInputDevice::CaptureCallback implementation.
   virtual void Capture(const std::vector<float*>& audio_data,
                        size_t number_of_frames,
-                       size_t audio_delay_milliseconds) OVERRIDE;
+                       size_t audio_delay_milliseconds,
+                       double volume) OVERRIDE;
   virtual void OnCaptureError() OVERRIDE;
 
   // AudioInputDevice::CaptureEventHandler implementation.
   virtual void OnDeviceStarted(const std::string& device_id) OVERRIDE;
   virtual void OnDeviceStopped() OVERRIDE;
 
   // webrtc::Module implementation.
   virtual int32_t ChangeUniqueId(const int32_t id) OVERRIDE;
   virtual int32_t TimeUntilNextProcess() OVERRIDE;
   virtual int32_t Process() OVERRIDE;
@@ skipping 117 matching lines @@
   virtual int32_t GetLoudspeakerStatus(bool* enabled) const OVERRIDE;
 
   // Sets the session id.
   void SetSessionId(int session_id);
 
   // Accessors.
   size_t input_buffer_size() const {
     return input_audio_parameters_.frames_per_buffer();
   }
   size_t output_buffer_size() const {
-    return input_audio_parameters_.frames_per_buffer();
+    return output_audio_parameters_.frames_per_buffer();
   }
   int input_channels() const {
     return input_audio_parameters_.channels();
   }
   int output_channels() const {
     return output_audio_parameters_.channels();
   }
   int input_sample_rate() const {
     return input_audio_parameters_.sample_rate();
   }
@@ skipping 57 matching lines @@
 
   // Protects |recording_|, |output_delay_ms_|, |input_delay_ms_|.
   mutable base::Lock lock_;
 
   int bytes_per_sample_;
 
   bool initialized_;
   bool playing_;
   bool recording_;
 
+  // Local copy of the current Automatic Gain Control state.
+  bool agc_is_enabled_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_